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It's not too late to register for the OpenSIPS Workshop @Cluecon Monday August 3, 2015
by Alex Goulis 29 Jul '15

29 Jul '15

There's still room for the OpenSIPS Workshop @ClueCon. The OpenSIPS Workshop is a new format that will focus more on providing attendees with more of a technical approach to integrating OpenSIPS rather than case studies - the sessions will provide working setups (with explanations) for OpenSIPS in different scenarios . The Workshops will cover new OpenSIPS release and features, FreeSwitch integration , Edge Proxy setup, Asynchronous I/O support and Fraud prevention. The Official schedule can be found at: http://www.opensips.org/Community/Workshop-2015Chicago-Schedule But we're happy to announce the following workshop leaders: Razvan Crainea - OpenSIPS project Dan Christian Bogos - ITsysCOM Vlad Paiu - OpenSIPS Project Eric Tamme - OnSIP Alex Goulis - Ratetel Pete Kelly - SourceVox Whether you are a beginner or a seasoned professional, the OpenSIPS Workshop @ClueCon has something for you. For beginners, get answers to questions like: * how to extend, expand and enhance FreeSwitch with OpenSIPS * intelligent load balancing of freeswitch servers * why OpenSIPS is the right choice for your network * Building a calling card solution using OpenSIPS and Freeswitch Seasoned professionals can: * Learn about new modules and features for the 2.x release * Discuss advanced topics * Discuss the development roadmap for OpenSIPS Once we've wrapped up the workshop, head over to the Hackathon with the OpenSIPS Team. The registration fees (per person) are: $199 for those attending only the OpenSIPS Workshop $149 for those attending ClueCon as well or having a discount code (*CLUECON2015*) or are registering 3 or more participants Space is limited, so don't wait too long to register. More information can be found at: http://www.opensips.org/Community/Workshop-2015Chicago We're looking forward to seeing you in Chicago! - The OpenSIPS Team

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G729 both in hardware and bypass
by Guillermo Ruiz Camauer 29 Jul '15

29 Jul '15

OK, this is probably not a usual setup, but I would like to know if it is possible: I have a FS server with 2 IP trunks configured. It also has a Digium hardware transcoding card which I use for 120 sessions of G729. I would like to set FS up in such a way that one of the IP trunks uses G729 in bypass mode. It is an outbound IVR and all voice prompts are pre-recorded in G729. I would like that the other trunk use the Digium transcoding hardware. Is there any way to configure this? Thanks in advance, -- Guillermo Ruiz Camauer

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freeswitch never seems to call sendmail when trying to email voicemails
by Jason Lewis 29 Jul '15

29 Jul '15

Hi, I've been trying to make freeswitch email voicemails but as far as I can tell, it never even calls sendmail. I have setting mailer-app to "sendmail" and "/usr/sbin/sendmail" to no avail. I can successfully send an email from the commandline using sendmail. (sendmail in this case is provided by postfix) I see no emails in the postfix mail logs when I leave a voicemail message. I also tried creating a shell just to see if it even gets called from fs, but it does not get called when a voicemail is deposited: #!/bin/bash echo $(date --rfc-3339=ns): $* >> /tmp/freeswitchsendmail.log After every change, I have run reloadxml and reload mod_voicemail. I have also tried restarting freeswitch. I am running the debian packages of FreeSWITCH Version 1.4.15-1~64bit (-1 64bit) my configuration is based on the vanilla configuration with only very minor changes. I'm at a loss as to how to debug further, but I'm pretty sure the mailer-app is never called. Is there some setting I'm missing or something obvious I'm not doing? My config: 1001.xml: <include> <user id="1001"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="1001"/> <param name="vm-mailto" value="jason(a)xyz.com.au"/> <param name="vm-mailfrom" value="vm(a)freeswitch.xyz.com.au"/>  </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1001"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1001"/> <variable name="effective_caller_id_number" value="1001"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> and in switch.conf.xml I have the following set: <param name="mailer-app" value="/usr/sbin/sendmail"/> <param name="mailer-app-args" value="-t"/> I made a log at level 7 and put it on the pastebin: https://pastebin.freeswitch.org/23921 Jason Lewis http://emacstragic.net

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[ERR] sofia_glue.c:313 Invalid tls-verify-policy value: none
by Kamil Nigmatullin 29 Jul '15

29 Jul '15

After fresh installation of 4.20 I keep getting this erroe sometimes 2015-07-29 08:54:37.376432 [ERR] sofia_glue.c:313 Invalid tls-verify-policy value: none Can I fix it somehow? Thank you -- Kamil Nigmatullin Tel: 77272323748 mob: 7 (707) 2517003 Skype: kamil.nigmatullin

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Ad hoc conference
by Yehavi Bourvine 29 Jul '15

29 Jul '15

Hello, It has been some time since the last time I wrote a dialplan for Freeswitch, so my mind is slightly rusted :-) I've been asked to do ad-hoc conferencing support for multi users conference: A calls B, puts him on hold, calls C, connects all together, put them on hold, call D and connects all together, and so on. This is the way they want it, like the old PBX we have. Is there some example of how to do such a thing? if not, I thought of the following: - When A presses *1 the first time, a conference room is created and the other party is transfered to it. - For the second and others: A initiates a new call, and when connected press *2 to connect the new party as well as A into the conference room, What I don;t know how to do is: How to take a bridged call, split it into two legs and move one or both into the conference room. Any idea will help. Thanks, __Yehavi:

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Implementing <Form> Tag of VXML in Freeswtich using Javascript
by david.fu＠oocl.com 28 Jul '15

28 Jul '15

Dear Freeswitch experts, I would like to implement something like <Form> and <submit> tag of VXML in Freeswtich using Javascript. The purpose is to sending request back to the application server from Freeswtich. However, I couldn't find the related API(s) in the Freeswitch official web site. Would you please help advise ? Thank you so much. Yours faithfully, David IMPORTANT NOTICE Email from OOCL is confidential and may be legally privileged. If it is not intended for you, please delete it immediately unread. The internet cannot guarantee that this communication is free of viruses, interception or interference and anyone who communicates with us by email is taken to accept the risks in doing so. Without limitation, OOCL and its affiliates accept no liability whatsoever and howsoever arising in connection with the use of this email. Under no circumstances shall this email constitute a binding agreement to carry or for provision of carriage services by OOCL, which is subject to the availability of carrier's equipment and vessels and the terms and conditions of OOCL's standard bill of lading which is also available at http://www.oocl.com.

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how to add mod_event_socket and mod_conference activities to the logfile
by Markus von Arx 28 Jul '15

28 Jul '15

Hallo I'd like to see more details about what's going on in the mod_event_socket and in the mod_conference modules in the FreeSWITCH log files. With the default (vanilla) configuration, nothing gets logged to the freeswitch.log logfile from inside these modules. I scanned all config files and the FS confluence wiki, but could not find any hint on how this can be achieved. The information I'd like to see in the freeswitch.log logfile is e.g. the following: * state changes in the mod_conference conferences, e.g. conference created, conference destroyed, member added, member removed, member muted, member unmuted, ... (including affected conference name and member id of course) * requests, replies & events in the mod_event_socket - if possible "shortened" and not including the whole event texts Any hints how this can be configured? Thanks a lot, Markus

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FreeSWITCH Week in Review (Master Branch) July 18th-July 24th
by Ken Rice 28 Jul '15

28 Jul '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1U48OzU FreeSWITCH Week in Review (Master Branch) July 18th-July 24th Hello, again. This passed week in the FreeSWITCH master branch we had 46 commits. The new features this week are: the addition of getcputime to retrieve FreeSWITCH process CPU usage, added support for 80 ms, 100 ms, 120 ms packetization to mod_opus, and added H.263 codec support to mod_av. Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-7848 [mod_opus] Add support for 80 ms, 100 ms, 120 ms packetization FS-7519 FS-7677 [mod_av] Add H.263 codec support FS-7885 Add getcputime to retrieve FreeSWITCH process CPU usage FS-7889 [mod_conference] Move conference chat to use an event channel so messages only go to the right ‘room’ for the conference and move conference chat functionality to use event_channel. Improvements in build system, cross platform support, and packaging: FS-7860 Prevent a switch_rtp header conflict FS-7130 Make /run/freeswitch persistent, so it will start under systemd The following bugs were squashed: FS-7789 [mod_av] Fixed issue with audio dropping out partway through recordings FS-7854 Add task_runtime to tasks table in core database FS-7856 [mod_av] Fix some segfaults and leaks. FS-7866 Fixed a crash when running incorrect var api expansion syntax “eval ${${external_sip_ip}:4}” FS-7861 FS-7862 [mod_conference] Fixed a crash and other issues caused by multi canvas feature FS-7681 [mod_conference] Factor out conference->canvas and allow per canvas record and play FS-7869 [mod_conference] Fixed a deadlock on shutdown after playing video file that will not display video FS-7654 Fixed regressions on eavesdropping on channels playing a file and on channels with unlike rates FS-7872 [mod_verto] Gracefully fail attempting to transfer 1 legged call FS-7874 [mod_conference] Fixed incorrect layout group count FS-7870 [mod_conference] Allow jsonapi commands to pass the string id field to pass special ID’s like “last” FS-7882 [mod_conference] Allow JSON API commands to send third arg for muting FS-7888 [mod_verto] Fixed namespacing problems in javascript library masked by global verto object FS-7811 Use more common format CIF for blank image FS-7902 [mod_local_stream] Fix for queue filling up when you have a mix of video and non video files FS-7891 [mod_spandsp] Allow spandsp dtmf detector to work on rates other than 8k FS-7839 Correct firefox > 38 DTLS behavior to match new EC requirements

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IP-Based Authentication
by Murugan Pandian 28 Jul '15

28 Jul '15

Hi, How i auth agent based on IP Address

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How can I have FS participate in a shared UDP multicast session/conference?
by D D 28 Jul '15

28 Jul '15

Hi, I want to have FS participate in a multicast session with other members. I see there is the esf_page_group application available, but I think it will constantly send rtp and therefor will monopolize the port.I think I would need to use VAD my call into esf_page_group. Is this the right approach? If so, how can I use VAD? Thanks,David

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